1. Field of the Invention
The invention relates to apparatus, and accompanying methods for use therein, for a telephony gateway intended for use, e.g., paired use, at opposite ends of a data network connection, in conjunction with at each end, e.g., a private branch exchange (PBX) for automatically routing telephone calls, e.g., voice, data and facsimile, between two peer PBXs over either a public switched telephone network (PSTN) or a data network, based on, among other aspects, cost considerations for handling each such call and called directory numbers, monitoring quality of service (QoS) then provided through the data network and switching (xe2x80x9cauto-switchingxe2x80x9d) such calls back and forth between the PSTN and the data network, as needed, in response to dynamic changes in the QOS such that the call is carried over a connection then providing a sufficient QoS.
2. Description of the Prior Art
Over the past century, telephone communications have become rather ubiquitous as the public switched telephone network (PSTN) has expanded into increasingly rural and other remote areas of the country, thus affording nearly universal telephone access. The PSTN provides real-time circuit-switched connections between caller and called parties, i.e., it establishes a continuous real-time link between caller and called locations, the latter often being specified by a string of digits entered by the caller; maintains that connection for the duration of a telephone call and then tears down that connection once that call terminates.
While basic plain old telephone service (POTs) connections typically provide continuous high quality analog connections, suited for voice, facsimile and relatively low speed data, such connections, based on their toll charges, can be expensive to use. Telephone companies frequently price these connections based on distance and time, i.e., a distance between the caller and called locations and duration of each call. Over the past few years in the United States, competition among regional and long distance telephone companies has existed and is intensifying, so much so as to effectively, in many instances, reduce telephone toll charges. However, such competition is only now emerging in many foreign countries. Further, various foreign governments have set relatively high interconnection tariffs to protect their local telephone companies, which are frequently governmentally regulated monopolies, from competitive pricing pressures arising from foreign carriers. Consequently, while telephone charges, on a per minute basis, are relatively inexpensive in the United States, the same is not true for telephone calls within and between foreign countries. In that regard, international calls between one country and another, such as the United States, can be rather expensive.
For many types of communication, such as data, continuous real-time switched connections, provided by the PSTN, are simply not necessary, given, e.g., relaxed latency restrictions for data, and are too costly.
Hence, within the last decade, private packet networks (commonly referred to as private xe2x80x9cdataxe2x80x9d networks) have experienced phenomenal growth as organizations, particularly those with computer and other digital equipment stationed at disparately located offices, sought cost-effective methods of communicating digital information between these offices. For ease of use and to accommodate as wide a universe of currently available network equipment and computer software as possible, these networks are generally designed to embody Internet Protocol (IP) based routing (which is the same methodology used in the Internet).
Though initial costs associated with implementing a private data network can be significant, average per use charges incurred through use of such a network tend to be considerably less than the toll charges for similar carriage, in terms of an amount of information being communicated, associated with the PSTN and hence, if the private network is sufficiently well used, can provide substantial cost savings to its owner as compared with equivalent use of the PSTN.
During the course of designing a private data network, various long-haul communication links that underlie the network are often chosen to provide bandwidth which, to accommodate anticipated growth, greatly exceeds current usage requirements. A common result of this is that many organizations, which have private data networks in operation, find themselves with significant amounts of unused (excess) installed bandwidth, which they have already built into their cost structure, available on their networks. Hence, some amount of additional traffic can be carried over this available bandwidth at what is, for all intents and purposes, essentially no additional cost. However, bandwidth is ephemeral: it is either consumed or not; it can not be stored for future use and hence, if not used when it is available, is simply wasted.
Those organizations that have implemented and use private data networks also tend to be extremely heavy telephone users as well, thus incurring substantial telephone charges on a regular continuing basis. These organizations include relatively large corporations, as well as government, academic and military organizations. Moreover, with increasing global computerization caused by explosive proliferation of personal computer usage over the past decade, even mid-sized and relatively small organizations with multiple offices are increasingly experiencing a need for access to an IP-based data network to facilitate inter-office data sharing and data communication. Though these organizations rarely, if ever, have sufficient usage to justify implementing their own networks, they are increasingly turning, for reasons of security and economy, to various network providers who offer secure access to a shared private IP network.
Recognizing the substantial telephone charges which these organizations regularly incur, particularly when viewed in the context of excess bandwidth available on their private data networks (whether dedicated or shared) and a near zero marginal cost of utilizing that bandwidth, these organizations would likely stand to economically benefit if this bandwidth could be used in some fashion to carry telephone calls that would otherwise be routed, at much higher cost particularly for international traffic, through the PSTN.
Currently, an effort, commonly referred to as xe2x80x9cVoice over IPxe2x80x9d (or more simply just xe2x80x9cVoIPxe2x80x9d), is underway in the art to develop technology and ultimately commercial products that can be utilized to transport, as an alternative to use of the PSTN, voice, data and facsimile communication, which would heretofore be carried over the PSTN, in packetized fashion over an IP data network, such as the Internet or a private data network. As currently envisioned in the art and described in A. Cary, xe2x80x9cIP PBXs: Open Questionsxe2x80x9d, Data Communications, March 1999, pages 69-83 and particularly page 72, products embodying this technology will probably utilize one of two basic approaches: (a) an xe2x80x9cadjunctxe2x80x9d approach, and (b) a LAN-based approach. The adjunct approach would use existing subscriber PBXs, subscriber line wiring and telephone sets but incorporate a VoIP telephony gateway, as an xe2x80x9cadjunctxe2x80x9d, at each of a number different sites. At each site, a corresponding gateway would be situated between PSTN trunk connections to a PBX at that site and connections to an IP network, so as to route incoming and outgoing telephone calls between PBX peers at these sites through the IP network. In contrast, the LAN-based approach would replace conventional telephone subscriber equipment and telephone PBXs with IP-compatible telephones to packetize voice calls, and carry these calls over local area networks (LANs).
The LAN based approach is likely to meet with significant disfavor and commercial skepticism owing to a substantial expense, particularly with large organizations that have extensive telephone systems, associated with removing and replacing existing telephone equipment, including PBXs and telephone instruments. This will be particularly true if, as we believe, the end-user price of a VoIP telephony gateway can kept to a reasonable level. Should this occur, the adjunct approach, by requiring a significantly reduced capital outlay while potentially providing substantial savings on telephone toll charges, will likely be widely adopted in the market and hence experience significant, widespread and rapid commercial success.
While carriage of telephony traffic over an IP network clearly holds theoretical promise and economic attraction, particularly through use of an xe2x80x9cadjunctxe2x80x9d approach, several obstacles exist, of which the following are illustrative. Any of these obstacles, if not properly addressed, could seriously hamper practical implementation and eventual deployment of this approach.
First, quality of service associated with a data connection provided through an IP network can vary widely. Such a connection can experience wide dynamic changes in latency, jitter and/or packet loss. Given the error correction processing that usually occurs at each end of a data connection, packet traffic can usually withstand transient changes, to a fairly significant degree, caused by any of these affects, before integrity of its payload data becomes jeopardized. However, voice traffic is particularly sensitive to these affects. Specifically, if packetized speech were to be subjected to transient changes in any of these affects, then this speech, once converted into an analog signal, may well contain audible distortion that might be highly objectionable to an individual on either end of a call. Consequently, any equipment that routes telephony traffic, originally destined to a PSTN, over an IP network instead must incorporate some mechanism to measure quality of service (QoS) of a networked connection, provided through the IP network, which carries telephony traffic and then switch this traffic over to the PSTN whenever the QoS of this connection sufficiently degrades. Preferably, this switchover itself should occur when distortion caused by a degradation in the QoS would likely become objectionable to a listener at either end of a call. In addition, this equipment should implement the switchover itself in a manner that is substantially inaudible, i.e., transparent, or at least not objectionable to that listener. In that regard, one illustrative device, referred to as xe2x80x9cSelsius-IP PBXxe2x80x9d gateway and recently developed by Cisco Systems, apparently switches a telephone call from the IP network to the PSTN should the latency on the IP network rise too far.
Second, not every telephone call needs to be routed over an IP network. In fact, no economic benefit results from routing certain calls over this network; these calls would best be handled through the PSTN. These calls include those which are, e.g., strictly local in nature, including, e.g., xe2x80x9c911xe2x80x9d calls, as well as those to toll-free numbers.
Furthermore, any VoIP gateway that is intended to carry telephone traffic must exhibit a very high degree of reliability and fault-tolerance, preferably similar to that of the PSTN itself.
Though efforts are currently underway at various organizations to develop a VoIP telephony gateway, between a PSTN and an IP network, to date, no commercial products appear to exist in the marketplace that implement IP telephony in a manner that remedies the above-noted obstacles.
In that regard, various gateways that have been announced seem deficient with respect to overcoming one or more of these obstacles. In that regard, one such illustrative device referred to as xe2x80x9cNetPhone IPBXxe2x80x9d gateway developed by NetPhone, Inc. of Marlborough, Mass. appears to provide a fallback capability to switch a telephone call to the PSTN from the IP network only in the event either the IP connection fails or a computer operating system, on which a software portion of the gateway executes, fails but not if QoS of the IP connection simply degrades.
No VoIP telephony gateway of which we are aware appears to be capable of selective call placement, i.e., deciding whether, from a nature of the telephone call itself, i.e., a called directory number, that call is best routed over the IP network or the PSTN and then routing the call accordingly, and/or is sufficiently reliable and fault-tolerant.
Therefore, a significant need currently exists in the art for a VoIP telephony gateway, particularly in view of the widespread adoption and substantial cost savings that could well accrue from its use, that is not only able to route a telephone call to an IP network in lieu of the PSTN but also can switch the call between these networks, as needed, based on QoS then being provided by the IP network. Such a gateway should also provide selective call placement such that those telephone calls that are not able to generate a cost savings, or other benefit, from being handled through the IP network are identified and routed to the PSTN rather than to the IP network. In addition, such a gateway should be highly reliable and fault-tolerant.
The present invention advantageously satisfies these needs, while overcoming known obstacles in the art, by providing a telephony gateway which, when operated with a similar peer gateway and each connected at an opposite ends of PSTN and data network connections, dynamically switches a call alternately between the data network and the PSTN based on real-time measurements of quality of service (QoS) then associated with the data network so as to carry the call over the particular network then providing sufficient QoS.
In accordance with our inventive teachings, once a telephone call has been initially routed to either the PSTN or to the data network (e.g., an IP network), then, should the QoS of a connection through the data network change, the call will be automatically switched (xe2x80x9cauto-switchedxe2x80x9d) to and routed through the other network, with the switching dynamically changing, during the duration of the call and in a substantially transparent manner to both the calling and called parties, alternating between the data network and the PSTN, as necessary, in response to dynamic changes in the QoS of the data network.
In particular, the inventive gateway determines network quality through dynamic measurements of latency, packet loss and error rate (jitter). Should either gateway involved in a call determine that network quality has either increased or decreased to necessitate an auto-switch either to the data network from the PSTN or the opposite, that gateway (hereinafter, for simplicity of reference, the xe2x80x9ccalling gatewayxe2x80x9d) will initiate an information exchange, using our inventive extensions to the H.323 protocol, with its peer gateway (hereinafter, the xe2x80x9ccalledxe2x80x9d gateway).
Specifically, if the call is to transition from the data network to the PSTN, the called gateway will select an available directory number from a pool of directory numbers (PDN) that has been assigned to it during its configuration and convey that specific number to the calling gateway. Once the calling gateway receives the particular PDN, it originates a circuit-switched call over its PSTN trunk connection to that PDN. The called gateway, sensing an incoming call on its PDN, will determine whether this number corresponds to the particular PDN on which that gateway is now expecting a call. If it is a different PDN number from that which it is expected, that gateway sends a message to the calling gateway over the network connection and waits for a gateway to claim this call. If this call is on the correct PDN, then the called gateway switches the call from its network connection to the now established circuit-switched connection through the PSTN. Once this occurs, the data network connection for this call is torn down by both gateways as if the call were completed. Auto-switching also occurs in reverse, from the PSTN back to the data network, when network quality sufficiently improves.
As per our inventive teachings, peered gateways facilitate auto-switching of telephone calls between the PSTN and the data network by establishing call-specific information for each call, including a unique call identifier (CallId), and Calling and Called Flags and communicating that information between themselves during call setup. Gateways communicate this information by embedding this information into various H.323 messages, specifically in a so-called xe2x80x9cnonstandard Dataxe2x80x9d field, using call independent signaling. By virtue of this information, the gateways on calling and called sides form the same association for each call routed therebetween and with a common CallId used for that call. This identifier distinguishes that call from any other then being handled by either gateway such that these two peered gateways, acting in unison, can switch this particular call between these networks, as needed, without affecting any other calls.
Specifically, through use of call independent signaling features of an H.323 standard, a Calling Flag is embedded within an H.323 SETUP message, and a Called Flag, a CallId and a selected PDN are all embedded within an H.323 CALL PROCEEDING or H.323 CONNECT message. In that regard, the contents of the Calling Flag, which are generated by a calling side, contains information, for a given call being established, which indicates, to a called side, whether, from the calling gateway, that call can be auto-switched. In response to this SETUP message, the called side generates and saves a CallId number which uniquely identifies that call and then passes that ID back to the calling side, along with the Called Flag and PDN. The Called Flag specifies whether, from the called gateway, this call can be auto-switched. The calling side then saves this information for later use in properly auto-switching the call between the data network and the PSTN, should a need to auto-switch then occur.
Our inventive gateway functions as an entity within an H.323 environment. The gateway implements at least one gatekeeper, to which the gateway registers itself, and at least one border element. The gatekeeper manages a group of endpoints which collectively constitute a zone. An administrative domain is formed of at least one gatekeeper and a border element connected to the gatekeeper(s) in the domain. The border element provides external network access into the administrative domain.
Advantageously, as a feature of our invention, for increased local redundancy, our inventive gateway also implements peered border elements. Peered border elements function together and behave as a single monolithic border element, i.e., one xe2x80x9clogicalxe2x80x9d border element, but with their functionality being duplicated across these such elements. Hence, if either of the peered border elements in an administrative domain fails, the other peered element can provide inter-domain routing and inter-zone routing within that domain. Peered border elements preferably have a loosely coupled distributed architecture, with no hierarchical differences. All transactions from gatekeepers or one border element in a domain are shared with its peer border element. As such, transaction data stored in one peered border element remains synchronized with that stored in the other, such that either one border element can immediately undertake transaction processing should its peer border element fail or be taken out of service.
Each peered border element has both TCP/IP server and client connections. Messages between peer border elements include information download and information update messages, as well as messages to establish and disconnect TCP/IP connections therebetween. An information download message is sent by one xe2x80x9coriginatingxe2x80x9d border element to its peer, upon establishing a TCP connection with that peer. This message shares all the call routing capabilities of the originating border element with its peer. The message contains local service relationships (internal to a domain), local descriptors, external service relationships (external to a domain) and external descriptors. The local service relationships define transport addresses of each of the gatekeepers that has a service relationship with the originating border element. The local descriptors define routing descriptors and are obtained from either a static configuration of the same domain as the originating border element or from gatekeepers, located within this domain, that have established service relationships with the originating border element. The external service relationships define, for the originating border element, transport addresses of those border elements external to this domain that have established a service relationship with the originating border element. The external descriptors define routing descriptors, that are obtained from either a static configuration of the H.323 environment or from border elements, located external to the domain that contains the originating border element, that have established service relationships with the originating border element. An information update message is sent from the originating border element to its peer in order to notify the latter of a change either in information affecting a gatekeeper located within the same domain or information received from a border element located external to this domain. The particular border element within an pair of xe2x80x9cpeeredxe2x80x9d border elements that originally received such information is responsible to send that information to all its peers.
Furthermore, each gateway advantageously provides, as another feature of our invention, selective call routing to route, based on called directory numbers, only those of its outgoing calls to the data network that can provide effective cost savings to the calling parties and/or their organizations. This routing is based on called number information, e.g., predefined called numbers and lists of bypass telephone numbers (BPN) and telephone exchanges, that can be programmed into the gateway during its configuration. As such, local calls and calls to xe2x80x9c911xe2x80x9d and the like which provide no appreciable cost savings, if any, to a calling party (or his(her) organization) are automatically routed to the PSTN for the entire duration of each such call.